1. Field of the Invention
The present invention relates to a hearing aid utilizing a digital signal processing technology.
2. Description of Related Art
Dysaudia or deafness can be generally classified into a conduction deafness and a perceptive deafness. The conduction deafness is such a condition that sound itself is not sufficiently transmitted because of abnormality of an external ear or middle ear. The conduction deafness can be satisfactorily compensated by a conventional analog hearing aid.
On the other hand, the perceptive deafness is such a condition that it is difficult to sense sound itself because of abnormality of an internal ear. This perceptive deafness is attributable to various causes, for example, lack of stereocilium at a tip end of frondose cells in a cochlea, or a trouble in nerve for transmitting a sound. A senile deafness is included in the perceptive deafness.
This perceptive deafness can be hardly overcome by the conventional analog hearing aid, and attention is being focused on a digital hearing aid capable of realizing a complicated signal processing.
In addition, the perceptive deafness exhibits various symptoms, each of which is greatly different from one person to another. One main symptom of the perceptive deafness includes a loudness recruitment phenomenon, in which a minimum level capable of hearing (minimum threshold of audibility) elevates, but a maximum level (maximum threshold of audibility) does not change much, with the result that an audible range is narrowed. This change is different from one frequency to another.
As a means for overcoming the above problem, it is a conventional practice to compress a dynamic range of an input sound, which is disclosed by, for example, Journal of Acoustic Society of Japan, Vol. 47, No. 10, pp 778-784, 1991 and Japanese Patent Application Laid-open Publication No. JP-A-3-284000.
Referring to FIG. 1, there is shown a block diagram illustrating the digital hearing aid disclosed by the first referred publication. The digital hearing aid shown in FIG. 1, which will be called a "first prior art example" in this specification, is so configured that an input signal obtained through an input 100 is divided by a low pass filter 102, a band pass filter 104 and a high pass filter 106 into three different frequency component, which are, in turn, analog-to-digital converted and distributed by a multiplexer and analog-to-digital converter 108 to a low frequency band gain setting circuit 110, a middle frequency band gain setting circuit 112 and a high frequency band gain setting circuit 114. Outputs of these gain setting circuits are supplied through digital-to-analog converters 116, 118 and 120 to smoothing filters 122, 124 and 126, respectively. Outputs of smoothing filters are combined by an adder 128, and then, supplied through an output limiter and power amplifier 130 to an output 132.
With this arrangement, an arbitrary input-to-output characteristics in each of the different frequency bands can be realized independently of the other frequency bands, so that an output signal confined within an arbitrary desired dynamic range is outputted for a hearing compensation.
Referring to FIG. 2, there is shown a block diagram illustrating the digital hearing aid disclosed by Japanese Patent Application Laid-open Publication No. JP-A-3-284000. The digital hearing aid shown in FIG. 2, which will be called a "second prior art example" in this specification, is so configured that an input signal obtained through an input 200 is analog-to-digital converted by an analog-to-digital converter 202, and then, supplied to a frequency sampling structure filter 204, whose output is digital-to-analog converted by a digital-to analog converter 206, and then, supplied to an output 208. Furthermore, an output of the analog-to-digital converter 202 is subjected to a short-time Fourier analysis in a short-time Fourier analyzing circuit 210, and Fourier coefficients obtained in the short-time Fourier analyzing circuit 210 are time-averaged by Fourier coefficient averaging circuits 212A, 212B, . . . , 212N. Averaged Fourier coefficients "a.sub.1 ", "a.sub.2 ", . . . "a.sub.n " are supplied to loudness mapping functions circuits 214A, 214B, . . . , 214N, respectively, which, in turn, output gains "g.sub.1 ", "g.sub.2 ", . . . "g.sub.n " required for the frequency sampling structure filter 204. Thus, a loudness is compensated for a hearing compensation.
As seen from the above, in order to compensate for the recruitment in the perceptive deafness, the hearing aid is required to convert an input signal, which varies in frequency and strength, into an output signal in matching with a hearing characteristics of a person to be fitted with the hearing aid. Therefore, a time-variant filter is used in the digital hearing aid to change the characteristics of the hearing aid in response to both the input signal and the hearing characteristics of the person to be fitted with the hearing aid.
In the first prior art example, however, since the input signal is divided into only three frequency bands, the hearing aid cannot meet with all different hearing characteristics of various deaf persons. In addition, if the three frequency band signals become out of phase, naturality of an outputted voice is deteriorated or lost.
On the other hand, since the second prior art example uses the frequency sampling structure filter as a hearing compensating filter, the following problems have been encountered. In the frequency sampling structure filter, a frequency component of the input signal is made small in the proximity of a "zero" but large at a "pole". This results in a drop of a S/N ratio, because of a calculation precision of a finite length.
Furthermore, in the case of changing the characteristics of the frequency sampling structure filter, it is necessary to change the filter coefficients at the same as a finite impulse response of the filter of the firstly set characteristics is ended. Otherwise, the impulse response changes in the way, so that the characteristics itself of the filter changes, with the result that a firstly determined characteristics cannot be found. Accordingly, at the time of changing the characteristics of the hearing compensating filter, it is necessary to monitor the impulse response of the filter or to perform the calculation of the impulse response, and therefore, the control becomes difficult.
In addition, since the frequency sampling structure filter has only control points distributed over a frequency with equal frequency intervals; the degree of freedom in design is low. In order to obtain a desired characteristics, it is necessary to increase the number of control points, namely to elevate the order of the filter, which results in an increased amount of calculation.